Synchronization and mixing of audio and video streams in network-based video conferencing call systems

ABSTRACT

In one aspect, audio streams are added to a mix until the mix is either complete (i.e., all audio streams have been added) or the mix is closed early (i.e., before the mix is complete). In another aspect, audio and video streams are synchronized by playing back the audio stream and then synchronizing display of the video frames to the playback of the audio stream.

CROSS REFERENCE TO RELATED APPLICATIONS

This application is a continuation application of Ser. No. 13/646,395,filed on Oct. 5, 2012, which claims priority to U.S. application Ser.No. 12/242,358, filed on Sep. 30, 2008 and U.S. Provisional ApplicationNo. 60/976,464, “Video Conference User Interface and Features” by MukundThapa filed on Sep. 30, 2007. All of these applications are incorporatedby reference herein in their entirety.

BACKGROUND OF THE INVENTION

Field of the Invention

The present invention relates generally to video conferencing over anetwork. In particular, the present invention is directed towardssynchronization and/or mixing of audio and video streams during anetworked video conference call.

Description of Background Art

Conventional networking software for video and audio conferencingpermits one-way, two-way and in some cases multi-way communicationbetween participants. Because each participant may be in a differentenvironment and at a different location on a network, the transmissionand reception of audio and video packets between various participantsand/or to a central server may vary among them. For example, aparticipant may receive packets from a nearby participant in a morereliable fashion and with less delay than those from a participant thatis more remotely located on the network. Packets may also be receivedout of order.

However transmitted and received over a network, audio and video datamust be synchronized and mixed during display in order to produce a goodvideo conferencing experience. For example, if the video and audio of aparticipant are not synchronized, then his mouth movements will notmatch his speech. The result can be annoying at best and can hindercommunications at worst. Similarly, if the audio and/or video ofdifferent participants are not synchronized, then the unexpected pausesand timing may be interpreted as hesitations or other gestures. This canalso hinder efficient communications between the participants.

Thus, there is a need for preferably simple approaches to synchronizingand mixing audio and/or video for networked participants in a videoconference call.

SUMMARY OF THE INVENTION

In one aspect, the present invention overcomes limitations of the priorart by adding audio streams to a mix until the mix is either complete(i.e., all audio streams have been added) or the mix is closed early(i.e., before the mix is complete).

In one approach, audio streams from N senders are to be mixed. The Naudio streams are received over a network. The audio streams are dividedinto portions that will be referred to as audio chunks (e.g., 40 msaudio chunks). The received audio chunks are buffered. A mix is openedand the process cycles through the N senders. If a sender's audio chunkhas not yet been added to the mix and it is available from the buffer,then the sender's audio chunk is added to the mix. If the sender's audiochunk is already in the mix and the sender has at least one additionalaudio chunk buffered (i.e., waiting for use in a future mix), a waitcounter is incremented for that sender. The mix is closed when audiochunks from all N senders have been added. It may also be closed earlyupon some predetermined condition based on the value of the waitcounter(s) (e.g., if the wait counter reaches a maximum value).

In a different approach, the process is driven by receipt of audiochunks. A mix is opened. As each sender's audio chunk is received, it isevaluated for inclusion in the mix. If the sender is not yet in the mixand the received audio chunk is the correct audio chunk for the mix,then it is added to the mix. Otherwise, it is buffered for a future mix.Again, the mix is closed if audio chunks from all N senders are in themix or if a predetermined early close condition is met. For example, aqueue counter may be used to count the number of audio chunks in eachsender's buffer. The mix may be closed early if the queue counterreaches some maximum value. In another aspect, once a mix is closed, theprocess attempts to use the audio chunks already stored in the buffersto create the next mix, rather than immediately creating a new mix basedon a newly received audio chunk.

Another aspect concerns synchronizing audio streams and video streams.In one approach, the audio stream is played as a series of audio chunks.The video stream is considered one frame at a time. A time marker forthe current video frame is compared to the expected time duration of thecurrent audio chunk. If the current frame should occur during thecurrent audio chunk, then it is displayed and the process moves to thenext frame. If the current frame should occur after the current audiochunk, then the process checks again later. If the current frame shouldhave occurred before the current audio chunk, then the frame isdiscarded and the process moves to a future frame.

These mixing and synchronization processes can be divided betweenclients and/or servers in different ways. For example, a client-serverarchitecture can be used where the server performs most of thefunctionality described above. Alternately, a client-server architecturecan be used where the server routes the various streams from client toclient but the clients perform most of the functionality describedabove. The functionality can also be split between client and server.Peer-to-peer architectures can also be used.

In a preferred approach, a central server receives audio and videostreams from each sender client. It sends the appropriate audio andvideo streams to each receiver client (recall that each client typicallywill act as both a sender client and a receiver client). Each receiverclient mixes the audio streams and synchronizes the mixed audio streamwith the video stream(s). In an alternate approach, the server mixes theaudio streams to produce the appropriate composite audio stream for eachreceiver client. The server sends to each receiver client the mixedaudio stream and any applicable video streams, and each receiver clientsynchronizes the received audio and video streams.

Other aspects of the invention include software, systems and componentsof systems for implementing the techniques described above. Yetadditional aspects include methods and applications for all of theforegoing.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram of a server-based architecture suitable foruse with the invention.

FIG. 2 is a screen shot of a participant's user interface for a videoconference.

FIG. 3 is a block diagram of an example client according to theinvention.

FIGS. 4-5 are flow diagrams of different methods for mixing audiostreams.

FIGS. 6A-6B are a flow diagrams of another method for mixing audiostreams.

FIG. 7 is a flow diagram of a method for synchronizing audio and videostreams.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

FIG. 1 is a block diagram of a server-based video conferencingarchitecture suitable for use with the invention. In this example, threeparticipants 102A-C are having a video conference. Each participant 102is operating a client device 110, which connects via a network 150 to acentral server 120. In this server-based architecture, the server 120coordinates the set up and tear down of the video conference and thecollection and distribution of audio and video streams from the clients110. In this particular example, each client 110 is a computer that runsclient software with video conferencing capability. To allow full videoand audio capability, each client 110 preferably includes at least onecamera (for video capture), display (for video play back), microphone(for audio capture) and speaker (for audio play back).

The clients 110 are connected via the Internet to the central server120. In this example, the central server 120 includes a web server 122,a call management module 124, an audio/video server 126 and anapplications server 128. The server 120 also includes user database 132,call management database 134 and audio/video storage 136. Theparticipants 102 have previously registered and their records are storedin user database 132. The web server 122 handles the web interface tothe clients 110. The call management module 124 and call managementdatabase 134 manage the video conference calls. For example, the callmanagement database 134 includes records of who is currentlyparticipating on which video conference calls. It may also includerecords of who is currently logged in and available for calls and/ortheir video conferencing capabilities. The audio/video server 126manages the audio and video streams for these calls. Streamingtechnologies, as well as other technologies, can be used. Storage ofaudio and video at the server is handled by audio/video storage 136. Theapplication server 128 invokes other applications (not shown) asrequired.

FIG. 2 is a screen shot of a participant 102's user interface for thevideo conference. In this example, there are three participants:Gowreesh, Alka and Lakshman. This is a multi-point example since thethree participants are at different network locations. However, theinvention can also be used for one-to-one situations (e.g.,two-participant video call) or with more participants. FIG. 2 showsGowreesh's screen as indicated by 200. The top-level control for theuser interface will be referred to as the main communicator element 210.It includes top level controls for video conferencing. These controlstypically are either displayed as graphical elements or implemented aspart of pull-down menus (or other similar user interface components).Controls can be implemented as buttons, tabs, toolbars, arrows andicons, for example.

The video conference is displayed in window 280. In this example, thewindow 280 displays video of the other two participants: Alka andLakshman. Gowreesh's audio system plays the corresponding audio.Ancillary window 290 lists the current participants and also providesfor text chat. Files can also be shared by clicking on the attachmenticon.

For purposes of explaining aspects of the invention, the participants102A-B and their clients 110A-B will be referred to as senders, andparticipant 102C and its client 110C will be referred to as thereceiver. In the example shown in FIG. 2, Alka and Lakshman are sendersand Gowreesh is the receiver. These terms are used because Alka andLakshman are sending audio and/or video data streams and Gowreesh isreceiving these data (or derivatives of them). Of course, in most videoconferences, participants will act as both senders and receivers,sending audio and video of themselves and receiving audio and video ofothers.

FIGS. 1-2 illustrate one example, but the invention is not limited tothese specifics. For example, client devices other than a computerrunning client software can be used. Examples include PDAs, mobilephones, web-enabled TV, and SIP phones and terminals (i.e., phone-typedevices using the SIP protocol that typically have a small video screenand audio capability). In addition, not every device need have bothaudio and video and both input and output. Some participants mayparticipate with audio only or video only, or be able to receive but notsend audio/video or vice versa. The underlying architecture also neednot be server-based. It could be peer-to-peer, or a combination ofserver and peer-to-peer. For example, participants that share a localnetwork may communicate with each other on a peer-to-peer basis, butcommunicate with other participants via a server. Other variations willbe apparent.

As described above, one challenge of network-based video conferencing isthat the various data streams from the senders 110A-B should besynchronized and mixed for display at the receiver 110C. In FIG. 2,Alka's audio and video should be synchronized to each other, andLakshman's audio and video should be synchronized to each other. Inaddition, Alka's and Lakshman's audio/video streams preferably shouldalso have some degree of synchronization. For example, if Alka asks aquestion, it is preferable that the video conference show Lakshmananswering with his actual timing (i.e., avoiding too much relative delayor advance). This requires some synchronization of Alka's and Lakshman'saudio and video streams. Alka's and Lakshman's audio streams typicallywould also be mixed together to form a composite audio stream forplayback to Gowreesh. These tasks can be made more difficult if each ofthese data streams is sent as packets over network 150 since timing isnot preserved in the transmission of packets. Some packets may propagatethrough the network 150 more quickly than others, thus arriving out oforder or not arriving at all.

In the following example, it will be assumed that each sender client110A-B creates the data streams for its respective participant 102A-B;that these data streams are sent to server 120 which retransmits them tothe receiver client 110C, and that the receiver client 110C isresponsible for synchronizing and mixing the data streams to produce theappropriate data streams for display to the receiver 102C. That is, inthis example, all synchronization and mixing are performed locally atthe client 110C.

This division of functionality is assumed primarily for purposes ofexplanation. In alternate embodiments, the functionality might bedivided in other ways. For example, some or all of the functionality canbe shifted from the receiver client 110C to the server 120. For example,the server (e.g., A/V server 126) might mix the audio streams to form acomposite audio stream and then send the composite audio stream and theoriginal video streams to the receiver client 110C. Alternately, theserver 120 might also mix video streams to form a composite video stream(e.g., one video stream that contains both Alka and Lakshman in FIG. 2)for transmission to the receiver client 110C. In these examples, theclient 110C may still be responsible for synchronizing received audioand video since transmission of packets over network 150 typically willnot preserve their timing. In another variant, the server 120 might alsosynchronize the audio stream and video stream, for example by combiningthe two data streams into a single data stream that contains both audioand video in the correct time relationship.

However, any architecture which shifts computational burden from theclients 110 to the server 120 will require more powerful servers and maylimit the scalability of the solution. For example, the mixing of videostreams at the server typically requires the server to decompress bothvideo streams, combine them (often into a non-standard format) and thenrecompress the mixed video stream. If a video conference has fourparticipants and each participant is viewing the three otherparticipants, this requires the server to decompress the four videostreams, combine them three at a time into four composite video streams,and then recompress the four composite video streams. If there aremultiple video conferences active at the same time, the burden on theserver scales accordingly and the server preferably would be sized tohandle the worst case computational burden. On the other hand, if thefunctionality is implemented in the clients, then the computationalresources available (i.e., the number of clients) naturally grows withthe number of participants and number of video conferences.

In a peer-to-peer architecture, each sender 110A-B might send its audioand video streams directly to each receiver 110C, which then isresponsible for locally synchronizing and/or mixing the various datastreams.

FIG. 3 is a block diagram of one example of a client for synchronizingand mixing audio and video streams according to the invention. Theclient includes audio buffers 310, audio stream decoders 320, audiomixer 330 and audio output module 340. The client also includes videobuffers 350, video stream decoders 360, optional video mixer 370 andvideo output module 380. The client receives audio streams 302 and videostreams 304 from the various sender clients 110A-B (via the server 120)and produces an output audio stream 392 (typically, only one) and outputvideo stream(s) 394 (possibly, more than one) for display on thereceiver client 110C. The output data streams are synchronized bysynchronization module 390. The input data streams usually will not bereceived in a synchronized manner.

Using FIG. 2 as an example, the audio stream 392 displayed by Gowreesh'sclient typically will mix the audio from Alka and Lakshman. The videostream 394 typically would include two video streams, one of Lakshmanand one of Alka. The audio and video streams 392, 394 are synchronized.

Consider first the mixing of different audio streams 302. Assume thataudio data is captured and played back in certain duration “audiochunks.” Currently, the capture is done in audio chunks of 40 ms each.The number of samples in each audio chunk is determined by the samplingfrequency (and possibly also the number of audio channels). These audiochunks are packetized and sent by the sender clients 110A-B to thereceiver client 110C. For simplicity, assume that an entire audio chunkfits into a single data packet. If multiple packets are required, thepackets can be reassembled into the original audio chunks.

When packets of audio are received over a network, there can be loss andalso delays. Thus, during mixing, for example, one sender's audio chunkmay be available but another sender's chunk may not be available as yet(but yet should be included in the mix to prevent distortion). In oneapproach, the idea is to cycle through the senders putting one audiochunk from each sender into the mix. If the process reaches a sender butthe sender's audio chunk is not available, then cycle through theremaining senders and, at the end of the cycle, come back and recheckwhether the sender's audio chunk is now available. The sender may berechecked a certain number of times before the process times out. In oneapproach, the existing audio chunks may be mixed by audio mixer 330without the missing audio chunks, which may be assumed as dropped.

FIGS. 4-6 are flow diagrams showing three different implementations formixing audio chunks. In these flow diagrams, audio chunk size isexpressed in milliseconds (ms). This will be the duration of audio thatwill be played before the next audio chunk is played. A “mix” is the setof all audio chunks that should be combined at a given instant. The mixmay have the audio chunks combined using standard approaches or may bekept separate for playback in a player which will mix it. If there aren+1 participants in a video conference, then there typically will be nsenders for each receiver. That is, the mix for the receiver at a time tshould include the audio chunks for time t from the n senders. Aparticular sender is “in the mix” if his audio chunk is available formixing. The mix is “complete” when all audio chunks are available formixing.

The following symbols are used in FIGS. 4-6. Senders are sometimesreferred to as users:

-   -   n is the number of audio streams that are to be mixed (i.e.,        number of senders). Typically, a value of n implies a video        conference with n+1 participants. A complete mix will have n        audio chunks, one from each sender.    -   user_is_in_mix is an array of dimension n. Each element k of the        array is either 0 or 1. If user_is_in_mix[k]=1, this means the        audio chunk for sender k is in the mix. A value of 0 means it is        not in the mix.    -   num_users_in_mix is the total number of senders currently in        the mix. This is the summation of the elements of the array        user_is_in_mix. If num_users_in_mix=n, then that mix is        complete. If <n, then it is incomplete.    -   wait_count_for_user is an array of dimension n.        wait_count_for_user[k] is the number of times that sender k, who        is already in the mix, has an audio chunk available for some        future mix, but must wait because the current mix is not yet        complete.    -   max_wait_count is the maximum value of wait_count_for_user for        any sender k before the mix is closed (even though still        incomplete). Analysis, confirmed by experimentation, suggests        that the value 3 works well, although other values can also be        used.    -   q_count_for_user is an array of dimension n.        queue_count_for_user[k] is the number of audio chunks that        sender k, who is already in the mix, has available for future        mixes. The audio chunks are queued because the current mix is        not yet complete.    -   max_q_count is the maximum value of queue_count_for_user for any        sender k before the mix is closed (even though still        incomplete).    -   k is a counter that counts through the senders.

Three example algorithms are described in FIGS. 4-6. In the first two,audio buffers are filled for each sender as packets arrive, and themixing process independently accesses these buffers. In the thirdexample, as each packet arrives, it is sent to the mixing algorithm andprocessed immediately if possible or else stored in a buffer for futureprocessing. The decoding of the packets is not directly relevant to thediscussion and can take place at one of several different points. Animportant concept in all the algorithms is the wait count or queuecount, which allows the handling of delays in when the packets arereceived.

The general idea behind FIG. 4 is as follows, with reference to FIG. 3.Audio chunks arrive over a network and are put into the appropriateaudio buffer 310, with different buffers 310 for each sender. Thistypically is an independent process and implemented as a separatethread. The mixing algorithm 330 is started 410, 415 independently andaccesses the audio buffers 310 in sequence (loop 470). For each audiobuffer (sender), if there is no audio chunk available 422, then theprocess proceeds 470 to the next audio buffer. If there is an audiochunk available 424, then the process checks 430 whether that sender isalready in the mix. If not 432, then the audio chunk is added 440 intothe mix (assuming the audio chunk is for the right time period). If asender is already in the mix 434, then his/her wait count is increased450 by 1. The process then checks 460 whether the mix should be closed.The mix is closed 464, 465, if the mix is now complete (i.e., allsenders are in the mix) or if the process meets some other predeterminedearly close condition, for example if the process times out or, in thiscase, if the maximum wait count for any sender is reached. If the mix isnot closed, the loop 470 increments to the next audio buffer. When thenext mix is opened 415, then as each sender's audio chunk is added 440to the mix, the wait count, if positive, is decremented (last step in440).

FIG. 5 is a variation of FIG. 4. The difference is that each time a newmix is opened 515, the wait count for all users is initialized to zero.Also compare step 540 to step 440.

FIGS. 4 and 5 typically are implemented as two threads because the audiochunks are received independently of when they are processed by themixing algorithm. FIG. 6A-6B is an example that coordinates thereceiving of audio chunks with the mixing. It can be implemented as asingle thread. In FIGS. 4 and 5, the process was driven by automaticallybuffering the audio chunks as they are received and then sequentiallycycling through the audio buffers. In FIG. 6, the process is driven bythe receipt of audio chunks.

Referring to FIG. 6A, the general idea is as follows. Audio chunksarrive over a network as mentioned before. This time, however, as eachchunk is received 610, it is evaluated for possible mixing. If a mix isnot 622 currently open, then a new mix is opened 640 and the receivedaudio chunk is added 650 to the mix (if for the correct time period). Ifa mix is already open 624, then there are two possibilities. If thissender is not 632 in the mix, then the audio chunk is added 650 to themix. If this sender is 634 in the mix, then the audio chunk is buffered660 for use in a future mix and the queue count for the user isincreased 660 by 1. In step 670, once each sender has an audio chunk inthe mix or the queue count reaches its maximum (or other early closecondition is met), the mix is closed 674, 675. Otherwise 672, theprocess waits to receive 610 the next audio chunk.

When a mix is closed 676, there may be several audio chunks in thebuffers (from step 660). If this is ignored, the buffers may overflow.Accordingly, in this example, when the mix is closed 675, a check 680 isperformed to see if the queue count of any sender is greater than zero.If not 682, then the process waits 610 to receive the next audio chunk.

However, if any queue count is greater than zero 684, then the processtries to use 690 these stored audio chunks. For example, a new mix couldbe opened in step 690 and any applicable stored audio chunks added tothe mix (which could be from more than one sender), decrementing thecorresponding queue counts. Various approaches can be used to do this.If the mix can be completed, then the process 680-690 repeats. Once theprocess 690 of trying to deplete the audio buffers is completed, theprocess returns to be driven by receiving 610 the next audio chunk. Theprocess of trying to use stored audio chunks can also be used in theprocesses of FIGS. 4-5.

FIG. 6B is a flow diagram of one approach to process 690. In thisexample, a new mix is opened 691. The process cycles 694 through thebuffers for the senders. If a sender has an audio chunk available 692,it is added to the mix 693 and the queue counter for that sender isdecremented. If audio chunks are available for all senders, then the mixwill be completed 695. In that case, the mix is closed 696. If any queuecount is greater than zero 697, then the process repeats. If the mix isnot complete, then the process returns to receive 610 the next audiochunk.

In FIG. 6, the queue count has a slightly different meaning than thewait count in FIGS. 4-5. In FIG. 6, the queue count for a sender is thenumber of audio chunks currently buffered waiting for a next mix. InFIGS. 4-5, the wait count was the number of times a particular senderhad to wait because he was already in the current mix and had additionalaudio chunks buffered for future mixes.

The above algorithms do not address where the mixed audio is stored.Typically the mix is stored in a buffer which is accessed by theplayback process. Thus, it may happen that when a new mix is opened, thebuffer may be full. In this case, one strategy is to check every few ms(for example S_(A)/8) if a slot is open in the buffer (due to playback).

Now turn to video synchronization. With respect to FIG. 2, Alka's videoshould be synchronized to Alka's audio. If Alka's and Lakshman's audiostreams have been mixed to produce a composite audio stream, then Alka'svideo should be synchronized to the composite audio stream. Audio-videosynchronization is preferably achieved by playing the audio stream andsynchronizing the video stream to the audio playback. This is due inpart because the audio stream has a tighter time tolerance (i.e., jittertolerance) for playback.

A time marker is added to each audio chunk or video frame captured. Inthe case of audio if a 40 ms audio chunk is captured, then the marker istracked as of the start of the audio sample. A 40 ms audio chunk,however, will have many audio samples. The exact number is determined bythe sampling frequency. Mixed audio streams also have time markers,preferably one for each sender's audio chunk in the mix. The originalaudio streams have time markers and, when they are mixed to form acomposite audio stream, the time marker preferably is retained for thecomposite audio stream. Note that the time marker need not be an actualtime stamp but can be any sort of relative counter.

The differences between the audio chunk versus video frames can beexplained in terms of how they are treated. For video, suppose 25 videoframes per second (fps) are captured. Then each video frame is displayedand held for 40 ms (1000/25). At 30 frames per second, each video frameis held for 33⅓ ms on display. For audio, suppose audio is captured in40 ms chunks. Then 40 ms worth of audio are played back at a time, butthat 40 ms audio chunk includes many audio samples per the samplingrate. The audio playback is effectively continuous relative to the videoplayback because there are many audio samples per video frame. Thus, thesynchronization problem is to match the video playback to the audioplayback. This can be done by suitably marking the two data streams andthen matching the marks within specified tolerances.

In some sense, the audio playback is used to clock the video playback.In one approach, synchronization occurs as follows.

-   -   If the time marker of the video frame matches the time of the        audio playback, then display the video frame.    -   If the time marker of the video frame is ahead of that for the        audio playback, then wait.    -   If the time marker of the video frame is behind that for the        audio playback, then skip the video frame.        The decision as to whether the video is behind, at, or ahead of        the audio is determined within a certain tolerance.

FIG. 7 is a flow diagram of a specific implementation, using thefollowing symbols:

-   -   S_(A) is the size of the audio chunk in milliseconds. Audio is        captured S_(A) ms at a time.    -   T_(A)[i] is the time at which the ith audio chunk was captured,        in milliseconds.    -   T_(V)[k] is the time at which the kth video frame was captured,        in milliseconds.    -   f is the frame rate, in frames per second.    -   f_(D) is the frame display duration, in milliseconds.        f_(D)=(1/f)*1000.    -   tol1 is a tolerance for the lower bound, in milliseconds. This        can be zero or higher. In practice, tol1=2 appears to work well        for S_(A)=40 ms.    -   tol2 is the tolerance for the upper bound. This can be zero or        higher. In practice, tol2=0 appears to work well.

In FIG. 7, the basic idea is that if T_(V)[k] falls within the timeperiod calculated for the current audio chunk, then video frame k shouldbe displayed. The nominal time period runs from T_(A)[i] toT_(A)[i]+S_(A). which starts at time T_(A)[i] and ends at timeT_(A)[i]+S_(A), Tolerances tol1 and tol2 are used to add robustness, sothat the calculated time period has a start time of T_(A)[i]−tol1 and anend time of T_(A)[i]+S_(A)+tol2. This assumes that the times T_(V)[k]and T_(A)[i] are measured relative to the same reference time. This canbe achieved, for example, by starting the audio and video capturethreads at the same time relative to a common clock. Alternately, thesender client can start the clocks for audio and video capture at thesame time. Equivalently, if the audio and video capture clocks usedifferent time references, the offset between the two can becompensated.

In more detail, the process initializes 710 by initializing the videoframe counter j and starting playback of the audio stream. In step 720,lower bound L and upper bound U are calculated for the current audiochunk being played. It is then determined 730 whether video frame jfalls within the time period spanned by the current audio chunk. If itdoes 735, then the video frame is displayed 750 and the counter j isincremented to move to the next video frame and the process is repeated725. If the video frame j occurs after 736 the current audio chunk(i.e., in the future), then nothing happens. The process waits 760 andrepeats 725 the process at a later time. If the video frame j was tohave occurred before 734 the current audio chunk, then the video frameis discarded 740 and the next video frame is tested 742 to see if itoccurs during the current audio chunk. This process can be repeateduntil the video stream catches up to the audio stream.

The present invention has been described in particular detail withrespect to a limited number of embodiments. One skilled in the art willappreciate that the invention may additionally be practiced in otherembodiments. First, the particular naming of the components,capitalization of terms, the attributes, data structures, or any otherprogramming or structural aspect is not mandatory or significant, andthe mechanisms that implement the invention or its features may havedifferent names, formats, or protocols. Further, the system may beimplemented via a combination of hardware and software, as described, orentirely in hardware elements. Also, the particular division offunctionality between the various system components described herein ismerely exemplary, and not mandatory; functions performed by a singlesystem component may instead be performed by multiple components, andfunctions performed by multiple components may instead performed by asingle component.

Some portions of the above description present the feature of thepresent invention in terms of algorithms and symbolic representations ofoperations on information. These algorithmic descriptions andrepresentations are the means used by those skilled in the art to mosteffectively convey the substance of their work to others skilled in theart. These operations, while described functionally or logically, areunderstood to be implemented by computer programs. Furthermore, it hasalso proven convenient at times, to refer to these arrangements ofoperations as modules or code devices, without loss of generality.

It should be borne in mind, however, that all of these and similar termsare to be associated with the appropriate physical quantities and aremerely convenient labels applied to these quantities. Unlessspecifically stated otherwise as apparent from the present discussion,it is appreciated that throughout the description, discussions utilizingterms such as “processing” or “computing” or “calculating” or“determining” or “displaying” or the like, refer to the action andprocesses of a computer system, or similar electronic computing device,that manipulates and transforms data represented as physical(electronic) quantities within the computer system memories or registersor other such information storage, transmission or display devices.

Certain aspects of the present invention include process steps andinstructions described herein in the form of an algorithm. It should benoted that the process steps and instructions of the present inventioncould be embodied in software, firmware or hardware, and when embodiedin software, could be downloaded to reside on and be operated fromdifferent platforms used by real time network operating systems.

The present invention also relates to an apparatus for performing theoperations herein. This apparatus may be specially constructed for therequired purposes, or it may comprise a general-purpose computerselectively activated or reconfigured by a computer program stored inthe computer. Such a computer program may be stored in a computerreadable storage medium, such as, but is not limited to, any type ofdisk including floppy disks, optical disks, CDs, DVDs, magnetic-opticaldisks, read-only memories (ROMs), random access memories (RAMs), EPROMs,EEPROMs, magnetic or optical cards, application specific integratedcircuits (ASICs), or any type of media suitable for storing electronicinstructions, and each coupled to a computer system bus. Furthermore,the computers referred to in the specification may include a singleprocessor or may be architectures employing multiple processor designsfor increased computing capability.

The algorithms and displays presented herein are not inherently relatedto any particular computer or other apparatus. Various general-purposesystems may also be used with programs in accordance with the teachingsherein, or it may prove convenient to construct more specializedapparatus to perform the required method steps. The required structurefor a variety of these systems will appear from the description above.In addition, the present invention is not described with reference toany particular programming language. It is appreciated that a variety ofprogramming languages may be used to implement the teachings of thepresent invention as described herein, and any references to specificlanguages are provided for disclosure of enablement and best mode of thepresent invention.

The figures depict preferred embodiments of the present invention forpurposes of illustration only. One skilled in the art will readilyrecognize from the following discussion that alternative embodiments ofthe structures and methods illustrated herein may be employed withoutdeparting from the principles of the invention described herein.

Finally, it should be noted that the language used in the specificationhas been principally selected for readability and instructionalpurposes, and may not have been selected to delineate or circumscribethe inventive subject matter. Accordingly, the disclosure of the presentinvention is intended to be illustrative, but not limiting, of the scopeof the invention.

I claim:
 1. A computer-implemented method for synchronizing an audiostream and a video stream, the method comprising: playing the audiostream as a series of audio chunks according to time markers for theaudio chunks; based on a comparison of a time marker for a current frameof the video stream to a calculated time period for playing of a currentaudio chunk, determining if the current frame of the video stream shouldoccur during the playing of the current audio chunk: if the currentframe should occur during the time period calculated for the currentaudio chunk, then displaying the current frame and moving to a nextframe; if the current frame should occur after the time period, thenwaiting; and if the current frame should have occurred before the timeperiod, then discarding the current frame and moving to a next frame,wherein the step of determining if a current frame of the video streamshould occur during the playing of the current audio chunk comprises:determining a time marker for the current video frame; determining astart time and an end time for the time period calculated for the audiochunk, the determining comprising: determining a nominal start time anda nominal end time for the audio chunk, adjusting the nominal start timeby a tolerance, and adjusting the nominal end time by another tolerance;and determining whether the time marker falls between the start time andthe end time.
 2. A computer program product for synchronizing an audiostream and a video stream, wherein the computer program product isstored on a non-transitory computer-readable medium that includesinstructions that, when loaded into memory, cause a processor to performa method, the method comprising: playing the audio stream as a series ofaudio chunks according to time markers for the audio chunks; based on acomparison of a time marker for a current frame of the video stream to acalculated time period for playing of a current audio chunk, determiningif the current frame of the video stream should occur during the playingof the current audio chunk: if the current frame should occur during thetime period calculated for the current audio chunk, then displaying thecurrent frame and moving to a next frame; if the current frame shouldoccur after the time period, then waiting; and if the current frameshould have occurred before the time period, then discarding the currentframe and moving to a next frame, wherein the step of determining if acurrent frame of the video stream should occur during the playing of thecurrent audio chunk comprises: determining a time marker for the currentvideo frame; determining a start time and an end time for the timeperiod calculated for the audio chunk, the determining comprising:determining a nominal start time and a nominal end time for the audiochunk, adjusting the nominal start time by a tolerance, and adjustingthe nominal end time by another tolerance; and determining whether thetime marker falls between the start time and the end time.
 3. Anon-transitory computer-readable medium that includes instructions that,when loaded into memory, cause a processor to perform a method forsynchronizing an audio stream and a video stream, the method comprising:playing the audio stream as a series of audio chunks according to timemarkers for the audio chunks; based on a comparison of a time marker fora current frame of the video stream to a calculated time period forplaying of a current audio chunk, determining if the current frame ofthe video stream should occur during the playing of the current audiochunk: if the current frame should occur during the time periodcalculated for the current audio chunk, then displaying the currentframe and moving to a next frame; if the current frame should occurafter the time period, then waiting; and if the current frame shouldhave occurred before the time period, then discarding the current frameand moving to a next frame, wherein the step of determining if a currentframe of the video stream should occur during the playing of the currentaudio chunk comprises: determining a time marker for the current videoframe; determining a start time and an end time for the time periodcalculated for the audio chunk, the determining comprising: determininga nominal start time and a nominal end time for the audio chunk,adjusting the nominal start time by a tolerance, and adjusting thenominal end time by another tolerance; and determining whether the timemarker falls between the start time and the end time.